Howdy, I'm trying to remove the media/rtp streams from an asterisk server for natted users so would like to know if this is possible with kamailio.
Qu's: What is the best option? rtpproxy/mediaproxy? nathelper?
If i use kamailio to achieve this does it mean that i still have to carry the rtp streams through the kamailio server instead?
Also will i need to change the logon info for the clients so they now logon to kamailio then i just point registrar to asterisk?
Can i use kamailio for sip trunks to asterisk and carry rtp and natted clients media streams rather than register to asterisk?
Many thanks, Taff..
Hello,
On 02/26/2009 12:59 AM, carl Lougher wrote:
Howdy, I'm trying to remove the media/rtp streams from an asterisk server for natted users so would like to know if this is possible with kamailio.
yes it is possible. nathelper+rtpproxy is the option I use and prefer because of flexibility and performances. You can see an example at: http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy
Qu's: What is the best option? rtpproxy/mediaproxy? nathelper?
If i use kamailio to achieve this does it mean that i still have to carry the rtp streams through the kamailio server instead?
through the rtpproxy server, which can be located on same or different machine than kamailio.
Also will i need to change the logon info for the clients so they now logon to kamailio then i just point registrar to asterisk?
Can i use kamailio for sip trunks to asterisk and carry rtp and natted clients media streams rather than register to asterisk?
Yes, you can register to kamailio, see registrar and usrloc modules.
Cheers, Daniel
Many thanks, Taff..
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Thanks for that. So does it mean by using rtpproxy you will therefore carry all the rtp streams through that server or can it be redirected to the sip provider from the endpoint?
Also how do you put the kamailio server in the equation? Do you set it up as an external proxy for the clients or do you register the clients to it then just use asterisk for the media/vmail etc?
--- On Thu, 26/2/09, Daniel-Constantin Mierla miconda@gmail.com wrote:
From: Daniel-Constantin Mierla miconda@gmail.com Subject: Re: [Kamailio-Users] Kamailio Newb questions To: c_lougher@yahoo.co.uk Cc: users@lists.kamailio.org Date: Thursday, 26 February, 2009, 9:16 AM Hello,
On 02/26/2009 12:59 AM, carl Lougher wrote:
Howdy, I'm trying to remove the media/rtp streams from an
asterisk server for natted users so would like to know if this is possible with kamailio.
yes it is possible. nathelper+rtpproxy is the option I use and prefer because of flexibility and performances. You can see an example at: http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy
Qu's: What is the best option? rtpproxy/mediaproxy? nathelper?
If i use kamailio to achieve this does it mean that i
still have to carry the rtp streams through the kamailio server instead?
through the rtpproxy server, which can be located on same or different machine than kamailio.
Also will i need to change the logon info for the
clients so they now logon to kamailio then i just point registrar to asterisk?
Can i use kamailio for sip trunks to asterisk and
carry rtp and natted clients media streams rather than register to asterisk?
Yes, you can register to kamailio, see registrar and usrloc modules.
Cheers, Daniel
Many thanks, Taff..
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-- Daniel-Constantin Mierla http://www.asipto.com
On 02/26/2009 01:19 PM, carl Lougher wrote:
Thanks for that. So does it mean by using rtpproxy you will therefore carry all the rtp streams through that server
yes, that is the role of RTPProxy - to proxy the RTP streams, therefore those go via the server.
If you want end-to-end RTP stream, then look at STUN, if the phones are not behind symmetric nat, it can help.
or can it be redirected to the sip provider from the endpoint?
Also how do you put the kamailio server in the equation? Do you set it up as an external proxy for the clients or do you register the clients to it then just use asterisk for the media/vmail etc?
I do everything in kamailio but the media services which i do with asterisk (vmail, ivr, ...) - authentication, registration, call routing is done in kamailio.
Cheers, Daniel
--- On Thu, 26/2/09, Daniel-Constantin Mierla miconda@gmail.com wrote:
From: Daniel-Constantin Mierla miconda@gmail.com Subject: Re: [Kamailio-Users] Kamailio Newb questions To: c_lougher@yahoo.co.uk Cc: users@lists.kamailio.org Date: Thursday, 26 February, 2009, 9:16 AM Hello,
On 02/26/2009 12:59 AM, carl Lougher wrote:
Howdy, I'm trying to remove the media/rtp streams from an
asterisk server for natted users so would like to know if this is possible with kamailio.
yes it is possible. nathelper+rtpproxy is the option I use and prefer because of flexibility and performances. You can see an example at: http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy
Qu's: What is the best option? rtpproxy/mediaproxy? nathelper?
If i use kamailio to achieve this does it mean that i
still have to carry the rtp streams through the kamailio server instead?
through the rtpproxy server, which can be located on same or different machine than kamailio.
Also will i need to change the logon info for the
clients so they now logon to kamailio then i just point registrar to asterisk?
Can i use kamailio for sip trunks to asterisk and
carry rtp and natted clients media streams rather than register to asterisk?
Yes, you can register to kamailio, see registrar and usrloc modules.
Cheers, Daniel
Many thanks, Taff..
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-- Daniel-Constantin Mierla http://www.asipto.com
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
So if i just want to hook off the rtp stream between endpoint and sip provider then am i better off to go for a stun server than nathelper/rtpproxy?
What are you using rtpproxy for that is different than stun?
--- On Thu, 26/2/09, Daniel-Constantin Mierla miconda@gmail.com wrote:
From: Daniel-Constantin Mierla miconda@gmail.com Subject: Re: [Kamailio-Users] Kamailio Newb questions To: c_lougher@yahoo.co.uk Cc: users@lists.kamailio.org Date: Thursday, 26 February, 2009, 12:27 PM On 02/26/2009 01:19 PM, carl Lougher wrote:
Thanks for that. So does it mean by using rtpproxy you
will therefore carry all the rtp streams through that server yes, that is the role of RTPProxy - to proxy the RTP streams, therefore those go via the server.
If you want end-to-end RTP stream, then look at STUN, if the phones are not behind symmetric nat, it can help.
or can it be redirected to the sip provider from the
endpoint?
Also how do you put the kamailio server in the
equation? Do you set it up as an external proxy for the clients or do you register the clients to it then just use asterisk for the media/vmail etc?
I do everything in kamailio but the media services which i do with asterisk (vmail, ivr, ...) - authentication, registration, call routing is done in kamailio.
Cheers, Daniel
--- On Thu, 26/2/09, Daniel-Constantin Mierla
miconda@gmail.com wrote:
From: Daniel-Constantin Mierla
Subject: Re: [Kamailio-Users] Kamailio Newb
questions
To: c_lougher@yahoo.co.uk Cc: users@lists.kamailio.org Date: Thursday, 26 February, 2009, 9:16 AM Hello,
On 02/26/2009 12:59 AM, carl Lougher wrote:
Howdy, I'm trying to remove the media/rtp streams
from an
asterisk server for natted users so would like to
know if
this is possible with kamailio.
yes it is possible. nathelper+rtpproxy is the
option I use
and prefer because of flexibility and
performances. You can see an
example at:
http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy
Qu's: What is the best option? rtpproxy/mediaproxy? nathelper?
If i use kamailio to achieve this does it mean
that i
still have to carry the rtp streams through the
kamailio
server instead?
through the rtpproxy server, which can be located
on same
or different machine than kamailio.
Also will i need to change the logon info for
the
clients so they now logon to kamailio then i just
point
registrar to asterisk?
Can i use kamailio for sip trunks to asterisk
and
carry rtp and natted clients media streams rather
than
register to asterisk?
Yes, you can register to kamailio, see registrar
and usrloc
modules.
Cheers, Daniel
Many thanks, Taff..
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-- Daniel-Constantin Mierla http://www.asipto.com
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-- Daniel-Constantin Mierla http://www.asipto.com
On 02/27/2009 12:07 AM, carl Lougher wrote:
So if i just want to hook off the rtp stream between endpoint and sip provider then am i better off to go for a stun server than nathelper/rtpproxy?
nathelper/rtpproxy does rtp relay on server, so the rtp goes: [caller] === [rtpproxy] === [callee]
In case of dealing with symmetric NAT, it is the only feasible way now to make it work.
If STUN is used, the rtp goes: [caller] === [callee] . But does not work for symmetric nat.
The best is to combine both of them so you get as less as possible RTP relaying on server.
Cheers, Daniel
What are you using rtpproxy for that is different than stun?
--- On Thu, 26/2/09, Daniel-Constantin Mierla miconda@gmail.com wrote:
From: Daniel-Constantin Mierla miconda@gmail.com Subject: Re: [Kamailio-Users] Kamailio Newb questions To: c_lougher@yahoo.co.uk Cc: users@lists.kamailio.org Date: Thursday, 26 February, 2009, 12:27 PM On 02/26/2009 01:19 PM, carl Lougher wrote:
Thanks for that. So does it mean by using rtpproxy you
will therefore carry all the rtp streams through that server yes, that is the role of RTPProxy - to proxy the RTP streams, therefore those go via the server.
If you want end-to-end RTP stream, then look at STUN, if the phones are not behind symmetric nat, it can help.
or can it be redirected to the sip provider from the
endpoint?
Also how do you put the kamailio server in the
equation? Do you set it up as an external proxy for the clients or do you register the clients to it then just use asterisk for the media/vmail etc?
I do everything in kamailio but the media services which i do with asterisk (vmail, ivr, ...) - authentication, registration, call routing is done in kamailio.
Cheers, Daniel
--- On Thu, 26/2/09, Daniel-Constantin Mierla
miconda@gmail.com wrote:
From: Daniel-Constantin Mierla
Subject: Re: [Kamailio-Users] Kamailio Newb
questions
To: c_lougher@yahoo.co.uk Cc: users@lists.kamailio.org Date: Thursday, 26 February, 2009, 9:16 AM Hello,
On 02/26/2009 12:59 AM, carl Lougher wrote:
Howdy, I'm trying to remove the media/rtp streams
from an
asterisk server for natted users so would like to
know if
this is possible with kamailio.
yes it is possible. nathelper+rtpproxy is the
option I use
and prefer because of flexibility and
performances. You can see an
example at:
http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy
Qu's: What is the best option? rtpproxy/mediaproxy? nathelper?
If i use kamailio to achieve this does it mean
that i
still have to carry the rtp streams through the
kamailio
server instead?
through the rtpproxy server, which can be located
on same
or different machine than kamailio.
Also will i need to change the logon info for
the
clients so they now logon to kamailio then i just
point
registrar to asterisk?
Can i use kamailio for sip trunks to asterisk
and
carry rtp and natted clients media streams rather
than
register to asterisk?
Yes, you can register to kamailio, see registrar
and usrloc
modules.
Cheers, Daniel
Many thanks, Taff..
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-- Daniel-Constantin Mierla http://www.asipto.com
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
-- Daniel-Constantin Mierla http://www.asipto.com
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Daniel,
I hate to open up an old and tired can of worms, but are there any recent performance tests that indicate whether rtpproxy or Mediaproxy 2.0 scales better? Mediaproxy 2.0 uses API calls into the kernel's IP forwarding subsystem now so I wonder if it can do better than rtpproxy running in userspace.
Like you, I much prefer rtpproxy - it's so much simpler and more flexible to deal with.
Just wondering.
-- Alex
Daniel-Constantin Mierla wrote:
Hello,
On 02/26/2009 12:59 AM, carl Lougher wrote:
Howdy, I'm trying to remove the media/rtp streams from an asterisk server for natted users so would like to know if this is possible with kamailio.
yes it is possible. nathelper+rtpproxy is the option I use and prefer because of flexibility and performances. You can see an example at: http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy
Qu's: What is the best option? rtpproxy/mediaproxy? nathelper?
If i use kamailio to achieve this does it mean that i still have to carry the rtp streams through the kamailio server instead?
through the rtpproxy server, which can be located on same or different machine than kamailio.
Also will i need to change the logon info for the clients so they now logon to kamailio then i just point registrar to asterisk?
Can i use kamailio for sip trunks to asterisk and carry rtp and natted clients media streams rather than register to asterisk?
Yes, you can register to kamailio, see registrar and usrloc modules.
Cheers, Daniel
Many thanks, Taff..
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Hi Alex,
On 02/26/2009 02:36 PM, Alex Balashov wrote:
Daniel,
I hate to open up an old and tired can of worms, but are there any recent performance tests
never compared the two of them. I used only rtpproxy so far and I was very happy, mainly regarding the development and community support. mediaproxy is in first place python, which slower than native C.
that indicate whether rtpproxy or Mediaproxy 2.0 scales better? Mediaproxy 2.0 uses API calls into the kernel's IP forwarding subsystem now so I wonder if it can do better than rtpproxy running in userspace.
Sometimes I prefer not to mess up with the kernel, I rather have an application crashed and restarted by monit or similar tool, than a kernel panic and full system crash. I do not know if using just kernel space is much faster as I haven't tested, however, there is a C alternative in SER (to be available soon via the http://sip-router.org project) for rtp relaying via kernel: http://cvs.berlios.de/cgi-bin/viewvc.cgi/ser/sip_router/modules/iptrtpproxy/
I would bet on it if I would get into rtpproxy limitations, so far didn't happened.
RTPProxy has the advantage of a public source repository, which is more convenient for me to deal with, as opposite to tarballs downloads. With a public repository I can track changes, report and fix issues easier.
To conclude, it might be just a matter of choice for rtp relay, but the nathelper module is much flexible and supported by many core developers of kamailio and ser.
Cheers, Daniel
Like you, I much prefer rtpproxy - it's so much simpler and more flexible to deal with.
Just wondering.
-- Alex
Daniel-Constantin Mierla wrote:
Hello,
On 02/26/2009 12:59 AM, carl Lougher wrote:
Howdy, I'm trying to remove the media/rtp streams from an asterisk server for natted users so would like to know if this is possible with kamailio.
yes it is possible. nathelper+rtpproxy is the option I use and prefer because of flexibility and performances. You can see an example at: http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy
Qu's: What is the best option? rtpproxy/mediaproxy? nathelper?
If i use kamailio to achieve this does it mean that i still have to carry the rtp streams through the kamailio server instead?
through the rtpproxy server, which can be located on same or different machine than kamailio.
Also will i need to change the logon info for the clients so they now logon to kamailio then i just point registrar to asterisk?
Can i use kamailio for sip trunks to asterisk and carry rtp and natted clients media streams rather than register to asterisk?
Yes, you can register to kamailio, see registrar and usrloc modules.
Cheers, Daniel
Many thanks, Taff..
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Daniel-Constantin Mierla wrote:
Hi Alex,
On 02/26/2009 02:36 PM, Alex Balashov wrote:
Daniel,
I hate to open up an old and tired can of worms, but are there any recent performance tests
never compared the two of them. I used only rtpproxy so far and I was very happy, mainly regarding the development and community support. mediaproxy is in first place python, which slower than native C.
I agree with everything you say, and understand that rtpproxy adheres much more closely to the Kamailio technology and political stack. :)
Although Mediaproxy is in Python, the relaying is not done in Python, so what I wonder is just objectively whether kernel-based relay is faster than userspace relay - even if in native C and in a very simple and lean binary.
On 02/26/2009 02:54 PM, Alex Balashov wrote:
Daniel-Constantin Mierla wrote:
Hi Alex,
On 02/26/2009 02:36 PM, Alex Balashov wrote:
Daniel,
I hate to open up an old and tired can of worms, but are there any recent performance tests
never compared the two of them. I used only rtpproxy so far and I was very happy, mainly regarding the development and community support. mediaproxy is in first place python, which slower than native C.
I agree with everything you say, and understand that rtpproxy adheres much more closely to the Kamailio technology and political stack. :)
Although Mediaproxy is in Python, the relaying is not done in Python, so what I wonder is just objectively whether kernel-based relay is faster than userspace relay - even if in native C and in a very simple and lean binary.
never tested, maybe someone else can comment. I will do it probably when rtpproxy in no longer enough :-) ...
Cheers, Daniel
-
Alex Balashov -
carl Lougher -
Daniel-Constantin Mierla